AudioCodes Mediant 2000 User Manual page 554

Parameter
]
[SIPForceRport]
Web: Reject Cancel after
Connect
CLI: reject-cancel-after-
connect
[RejectCancelAfterConn
ect]
Web: Verify Received
RequestURI
CLI: verify-rcvd-requri
[VerifyReceevedReques
tUri]
Web: Max Number of
Active Calls
EMS: Maximum
Concurrent Calls
[MaxActiveCalls]
Web: QoS statistics in SIP
Release Call
[QoSStatistics]
User's Manual
The valid value range is 1 to 50. The default is 50.
Determines whether the device sends SIP responses to the UDP port
from where SIP requests are received even if the 'rport' parameter is not
present in the SIP Via header.

[0] = (Default) Disabled. The device sends the SIP response to the
UDP port defined in the Via header. If the Via header contains the
'rport' parameter, the response is sent to the UDP port from where
the SIP request is received.

[1] = Enabled. SIP responses are sent to the UDP port from where
SIP requests are received even if the 'rport' parameter is not present
in the Via header.
Determines whether the device accepts or rejects a SIP CANCEL
request received after the receipt of a 200 OK, during an established
call.

[0] = (Default) Accepts the CANCEL, by responding with a 200 OK
and terminating the call session.

[1] = Rejects the CANCEL, by responding with a SIP 481
Call/Transaction Does Not Exist, and maintaining the call session.
Enables the device to reject SIP requests (such as ACK, BYE, or re-
INVITE) whose user part in the Request-URI is different from the user
part received in the Contact header of the last sent SIP request.

[0] Disable = (Default) Even if the user is different, the device
accepts the SIP request.

[1] Enable = If the user is different, the device rejects the SIP
request (BYE is responded with 481; re-INVITE is responded with
404; ACK is ignored).
Defines the maximum number of simultaneous active calls supported by
the device. If the maximum number of calls is reached, new calls are
not established.
The valid range is 1 to the maximum number of supported channels.
The default is the maximum available channels (i.e., no restriction on
the maximum number of calls).
Enables the device to include call quality of service (QoS) statistics in
SIP BYE and SIP 200 OK response to BYE, using the proprietary SIP
header X-RTP-Stat.

[0] = Disable (default)

[1] = Enable
The X-RTP-Stat header provides the following statistics:

Number of received and sent voice packets

Number of received and sent voice octets

Received packet loss, jitter (in ms), and latency (in ms)
The X-RTP-Stat header contains the following fields:

PS=<voice packets sent>

OS=<voice octets sent>

PR=<voice packets received>

OR=<voice octets received>

PL=<receive packet loss>

JI=<jitter in ms>

LA=<latency in ms>
Below is an example of the X-RTP-Stat header in a SIP BYE message:
554
Description
Document #: LTRT-68822
Mediant 2000